CN101305588A - Network support for enhanced voip caller id - Google Patents
Network support for enhanced voip caller id Download PDFInfo
- Publication number
- CN101305588A CN101305588A CNA2006800417507A CN200680041750A CN101305588A CN 101305588 A CN101305588 A CN 101305588A CN A2006800417507 A CNA2006800417507 A CN A2006800417507A CN 200680041750 A CN200680041750 A CN 200680041750A CN 101305588 A CN101305588 A CN 101305588A
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- Prior art keywords
- calling
- call
- name
- phone
- terminal
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/42025—Calling or Called party identification service
- H04M3/42034—Calling party identification service
- H04M3/42042—Notifying the called party of information on the calling party
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
- H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
- H04M7/1225—Details of core network interconnection arrangements
- H04M7/1235—Details of core network interconnection arrangements where one of the core networks is a wireless network
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- Engineering & Computer Science (AREA)
- Computer Networks & Wireless Communication (AREA)
- Signal Processing (AREA)
- Telephonic Communication Services (AREA)
Abstract
An apparatus in one example has: a calling terminal and a called terminal in a telecommunication system; at least one of a specified name or a preferred call back number that are input at the calling terminal before originating a call to the called terminal; at least one telecommunication system that operatively couples the calling terminal to the called terminal; and the specified name and/or the preferred call back number being calling party information at the called terminal for the call from the calling party.
Description
Technical field
The present invention relates generally to communication network, and relate more specifically to have the communication network of caller ID (caller ID) feature.
Background technology
Wireless and wired communication system is constantly developing.System designer is developed the more feature of big figure for service provider and terminal use constantly.In the radio telephone system field, very big progress had been arranged in recent years based on the telephone system of honeycomb.Radio telephone system is based on various modulation techniques and available, and can use a plurality of frequency bands that distribute.Available modulation scheme comprises analog fm (FM) and uses the digital modulation scheme of time division multiple access (TDMA) and code division multiple access (CDMA).Every kind of scheme has the intrinsic merits and demerits relevant with system architecture, channeling and communication quality.Yet manufacturer offers the feature of service provider and feature that service provider offers the user is similar between different wireless systems.
No matter employed modulation scheme is how, the terminal use can with radio telephone all have a plurality of key characters.Nearly all radio telephone comprise the keyboard that is used for input digit and text at least and allow user's videotex, dial the number, the display of picture and incoming call person's number.In addition, radio telephone can comprise electric phone book, speed dialling, button voicemail access and such as the message transfer capability of Email.
Useful especially feature provides the caller ID in radio telecommunications system and the line telecommunications system.Caller ID be allow the recipient of incoming call in addition before replying, determine ongoing incoming call from the Network feature of number.
Except known wired and radio telecommunications system, also there is Internet telephony now.Internet telephony is to make people can use the class hardware and software of internet as the transmission medium of call.For the user that the internet with free or regular price inserts, the Internet phone software provides cheap call basically in the world Anywhere.The Internet telephony products is sometimes referred to as IP (Internet Protocol) phone, internet voice-bearer (VOI) or Voice over IP (VoIP) product.VoIP or voice over the Internet protocol are the processes that sends voice telephone signals by internet or other data network.If this telephone signal is analog form (voice or a fax), be digital form at first then with this conversion of signals.Add the grouping routing iinformation to this audio digital signals then, so that can pass through internet or other this audio digital signals of data network route.
Nowadays, telecommunication system show Calls side's message context when calling out by voip network has many problems.Thereby, there is the demand that is provided at the caller ID feature of moving in the Internet telephony environment in the prior art.
Summary of the invention
A kind of embodiment comprises a kind of equipment.This equipment can comprise: calling terminal in the telecommunication system and terminal called; In appointment name of before initiating, importing or the preferred ans back at least one in the calling terminal place to the calling of terminal called; At least one telecommunication system that in operation, calling terminal and terminal called is coupled; And described appointment name and/or described preferred ans back are the caller information from the calling of calling party at the terminal called place.
Another embodiment comprises a kind of equipment.This equipment can comprise: initiate the calling terminal to the voip call of the terminal called in the telecommunication system; In appointment name of before initiating, importing and/or the preferred ans back at least one in the calling terminal place to the voip call of terminal called; At least one telecommunication system that in operation calling terminal and terminal called is coupled, this telecommunication system have data network and have in PTSN (PSTN) or the cellular network at least one; And described appointment name and/or described preferred ans back are the caller information from the voip call of calling party at the terminal called place.
A kind of embodiment comprises a kind of method.The embodiment of this method can comprise: at the calling terminal place, and at least one before initiating in designated call person name or the preferred ans back to the calling of the terminal called in the telecommunication system; Initiate this calling and route the call to terminal called; And provide described name and preferred ans back at the terminal called place as caller information.
Another embodiment comprises a kind of method.The embodiment of this method can comprise: before the calling of initiating the terminal called in the telecommunication system, in designated call person name or the preferred ans back different with the acquiescence ans back of calling terminal at least one specified; And the person's name that provides the designated call at the terminal called place and preferred ans back are as caller information.
Description of drawings
From specification, claim and accompanying drawing, the feature of exemplary embodiment of the present invention will become apparent, in the accompanying drawings:
Fig. 1 is the expression of a kind of embodiment of the equipment of communication network with caller ID feature;
Fig. 2 is the expression that is used for providing at communication network an example flow chart of caller ID feature functionality;
Fig. 3 is the expression that is used for providing at communication network another example flow chart of caller ID feature functionality.
Embodiment
A kind of method of this method and equipment is to initiate voip call or PSTN common calling or office call (office call) the caller to specify name and preferred ans back for the caller before.The another kind of method of this method and equipment is to make network by VoIP and specified caller's name and the preferred ans back of PSTN network route.The another kind of method of this method and equipment is that the callee presented in specified caller's name and preferred ans back.The embodiment of this method and equipment can be with multiple network, such as the use that combines of PTSN network, data network, cellular network and this network.The embodiment of this method and equipment is particularly useful to the calling of being undertaken by the internet.
Fig. 1 has a kind of expression of embodiment of the equipment of caller ID feature according to this method and equipment, communication network.VoIP be call pass through the internet, such as the transmission of IP network 114.The internet sends little packet by packet switching on network.At Yuan Chu, lot of data is divided into many groupings.Each grouping is given tells network that each grouping is routed to address where.At the target place, described grouping is reassembled as primary data.Packet switching is very effective, thereby must keep the time quantum of connection between two sources and reduce load on the network because it minimizes.
For IP phone or VoIP, in case started session, just will be converted to number format from speaker's analog voice signal, with this Signal Compression the IP grouping that transmits by the internet.At the receiving terminal place, this signal is condensed to digital signal from the grouping decompress(ion), for the listener this digital signal is converted to analog signal subsequently.Voip call can take place in all cases.In order to call out, the user must be able to use and comprise wideband transmit standard, gateway and the assembly of necessity of adapter in some cases.
Have several different VoIP situations, comprise computer to computer, computer to phone, phone to computer and phone to phone.Though call out and follow identical basic format, depend on the source and target of speech data, aspect transmission, there are some differences.
From a computer to another computer, for example PC (personal computer) 116 require each computer user all to have identical software, microphone, loud speaker, sound card and high-speed Internet with calling that PC 118 is carried out to be connected.When computer is called out, do not relate to gateway.VoIP software will be mapped to recipient's computer and start session.As the part of session, will between computer, realize two channels, channel of each direction.This means that each computer all knows the packet of expectation from another computer.
When calling out by computer, this computer sends to recipient's computer with the analog voice signal digitlization with its boil down to grouping and with it by the internet.This grouping of recipient's computer organization and their decompress(ion)s are condensed to primary data is so that the recipient listens to.Identical thing takes place from recipient to caller.When end of conversation, caller's computer will send the signal that stops session to recipient's computer.
Do not exist charge from a computer to the trunk call that another computer carried out; Yet, may have every month ISP (the Internet service provider) expense of the connection of internet.
From the computer to the phone, the calling carried out of for example PC 118 and user's residence phone 106 requires the computer user to have essential software, microphone, loud speaker, sound card and high-speed Internet to be connected.This computer software will be mapped to gateway, such as with recipient's PSTN, such as PSTN and SS7 102 immediate voip gateways 112.In case set up session, just between computer 118 and gateway 112, realized two channels.This means that computer 118 and gateway 112 know that expectation is from each other packet.On the PSTN102 between gateway 112 and the recipient 106, also will there be open electric circuit.
When calling out by computer 118, computer 118 sends to gateway 112 with its boil down to grouping and with it by internet 114 with the analog voice signal digitlization.Gateway 112 is organized this grouping and their decompress(ion)s is condensed to primary data, so that pass to recipient 106.When data during from recipient 106, gateway 112 is the analog voice signal digitlization, with its boil down to IP grouping, and it moved on the internet 114, so that send caller's computer 118 to.When grouping was received by computer 118, grouping is organized and is decompressed was primary data.When end of conversation, caller's computer 118 will send the signal that stops session to gateway 112.Gateway 112 will be closed the circuit between it and the recipient 106 subsequently.Recipient 106 can freely accept other calling now.In case this session is terminated, gateway 112 is deleted the mapping of this gateway to computer from its memory.This class is called out may be had by caused little per minute expense of the connection between gateway 112 and the recipient 106 and the cost of being collected by gateway 112, but will be than traditional trunk call considerably cheaper.In order to save cost, it is approaching that employed PSTN 102 is positioned at recipient 106 as far as possible, so that minimize the price that gateway connects to the recipient.Every month ISP expense that may have equally, the connection of internet 114.
By phone, the calling carried out such as user's residence phone 106 and computer, such as PC 118 will have only when the caller at first dial in to gateway, during such as gateway 112 and have only when the computer user has and just work when essential software, microphone, loud speaker, sound card are connected with high-speed Internet.After caller 106 is connected to gateway 112, dial recipient 118 number.This number is stored by gateway 112 temporarily.Gateway 112 is checked the form of the number of being imported and is determined whom this number is mapped to subsequently.In when mapping, this number is appended on recipient's the IP address of computer 118.In case set up session, between gateway 112 and computer 118, realized two channels.This means that gateway 112 and computer 118 know that expectation is from each other packet.On the PSTN 102 between caller 106 and the gateway 112, also will there be open electric circuit.
When by gateway 112 when recipient 118 calls out, gateway 112 is the analog voice signal digitlization, moves on the internet 114 with its boil down to IP grouping and with it, so that send computer 118 at the receiving terminal place to.When grouping was received by computer 118, grouping is organized and is decompressed was primary data.When data during from recipient 118, computer 118 sends it to gateway 112 with the analog voice signal digitlization with its boil down to grouping and by internet 114.Gateway 112 is organized this grouping and their decompress(ion)s is condensed to primary data so that pass to caller 106.
When the caller finishes talk and hangs up the telephone 106 the time circuit between shutdown call person 106 and caller's the gateway 112.In case closed this circuit, this caller's telephone wire can freely be accepted other calling.Gateway 112 sends the signal that stops session to recipient's computer 118 subsequently.In case stopped this session, gateway 112 is just deleted the mapping of this gateway to computer from its memory.
This class is called out and will only be born by to caused local call cost of the connection of gateway 112 and the cost collected by gateway 112.The computer user may have to the internet every month ISP expense of 114 connection.
VoIP is used to the business that route might be derived from and end at traditional pstn telephone.Calling from a phone 106 to another phone 126 is with the connection to gateway 112.Before place calls person wishes the personnel that get in touch and local number, must at first dial special number so that contact gateway 112.After on being connected to gateway 112, caller 106 dials him or she and wants the side's that talks number and this number to be stored by gateway 112 temporarily.Gateway 112 is checked the form of the number of being imported and is determined whom this number is mapped to subsequently.When mapping, this number is appended on the IP address of another gateway 122.This another gateway 122 is directly connected on the PSTN 124 of the number of being dialed or as far as possible near this PSTN 124.As the part of session, will between gateway 112 and 122, realize two channels, channel of each direction.This means that each gateway knows the packet of expectation from another gateway.
When by gateway 112 when recipient 126 calls out, gateway 112 is the analog voice signal digitlization, with its boil down to IP grouping, and it moved on the internet 114 so that send gateway 122 at the receiving terminal place to.When grouping was received by gateway 122, grouping was organized and is decompressed for primary data and be delivered to recipient 126.106 identical thing takes place from recipient 126 to the caller.Hold 106 gateway 112 to keep circuit open between its own and caller 106 the caller.Hold 126 gateway 122 to keep circuit open between its own and recipient 126 the recipient.These open electric circuits are to be connected with 122 PSTN (such as PSTN 102 and 124) to gateway 112.
When caller 106 finished talk and hangs up the telephone, the circuit between caller 106 and caller's gateway 112 was closed.In case this circuit is closed, caller's telephone wire just can freely be accepted other calling.Gateway 112 sends the signal that stops session to recipient's gateway 122 subsequently.Be closed in circuit between it and the recipient 126 at the gateway 122 at recipient's end place.Recipient 126 also can freely accept other calling now.In case stopped this session, gateway 112 and 122 is just deleted the mapping of this number to gateway from memory.
The expense of the voip call between the phone is more much lower than traditional trunk call.The unique cost relevant with the voip call of carrying out between two phones is caused local call cost and any cost of being collected by the gateway network operator when getting in touch gateway.Gateway at recipient's end will more or less be charged according to the distance between this gateway and recipient's the telephone system.
Can handle calling in a similar fashion from miscellaneous equipment.For example, can initiate or receipt of call by the user's mobile phone 132 that is connected to by base station 130 and mobile switching centre 128 on the PSTN 124.Also can make a call by the public telephone office (PCO) that is coupled on the PSTN 102 by pay phone 100,112,114.For the phone of voip phone can be directly and internet or IP network 114 couplings.
Call card system 104 can be connected on the PSTN 102.Call card or phonecard can be the prepaid card or the credits card that can be used to pay the expense of call.The virtual call card also is known, and typically immediately for you provide access code at line service, but the actual call card that is similar to no ticket charging (ticket-less billing) is unknown.Correspondingly, it is a kind of calling that call card is called out, and is not to the telephone number charging initiated but the call card charging to being issued for this purpose by Local Exchange or long distance phone company at this callings.
In general, call card is the prepaid service phone that does not have monthly cost.Utilize call card, can use any phone even public telephone to make a call anywhere.Each call card has the number that is called as the PIN number, and this number is to use this business required.PIN is the specified personal identification number of this particular telephone card.
Number for access is in order to enter the telephone number that the call card system is dialed.In the U.S., Number for access is 800 free codes normally, and this number places the user on the call card network and allows the user to carry out cost-effective calling.Usually, Number for access is found at the back side of phonecard.
Fig. 2 is the expression that is used for providing at communication network an example flow chart of caller ID feature functionality.This embodiment of this method can comprise: at the calling terminal place, and before the calling of initiating the terminal called in the telecommunication system, at least one (201) in designated call person name or the preferred ans back; Make a call and route the call to terminal called (202); Provide name and preferred ans back as caller information (203) at the terminal called place.
Fig. 3 is the expression that is used for providing at communication network another example flow chart of caller ID feature functionality.This embodiment of this method can comprise: before the calling of initiating the terminal called in the telecommunication system, in designated call person name or the preferred ans back different with the acquiescence ans back of calling terminal at least one specified (301); And provide specified caller's name and preferred ans back at the terminal called place as caller information (302).Especially:, when call card is activated, designated call person name and preferred ans back are specified (311) for call card; For the calling of being undertaken by public telephone office, pstn telephone has the specified caller's name of before making a call input and the optional feature (312) of ans back; For the calling of being undertaken by public telephone office, public pstn telephone has the optional feature (313) of specified caller's name of before making a call input and preferred ans back; Call out to PSTN for public telephone office PC, before making a call, pc client is caught specified caller's name and preferred ans back (314); And for the calling of being undertaken by the residence voip phone, as the part of activation of service, the person's name that provides the designated call and preferred ans back (315).
About IP network, IP network can be used such as the new information in SIP (session initiation protocol), the signaling protocol that H.323 waits and come route caller name and ans back.In case this message arrives voip gateway, voip gateway just can use this information to fill the PSTN caller information, rather than is put in its information.SIP is to use text format messages to set up, manage and stop the application layer protocol of multimedia communication sessions.SIP is the H.323 simple version of packet multimedia system of ITU.SIP is defined in RFC 2543.
Terminal called can use the current techniques that is used to show current caller information to come show Calls side to strengthen information.Calling party strengthens information and can show with the multiple form such as visual and/or audio frequency.
Present device in example can comprise a plurality of assemblies, one or more such as in electronic building brick, nextport hardware component NextPort and the computer software components.A plurality of such assemblies can be combined in this equipment or divide.
Step as described herein or operation are exemplary.Under the situation that does not depart from spirit of the present invention, can there be many flexible programs of these steps or operation.For example, described step can be carried out by different orders, perhaps can add, deletion or modify steps.Although at length described and described exemplary embodiment of the present invention here, but will be it is evident that for those skilled in the relevant art, can carry out various modifications, increase, substitute etc., and do not depart from spirit of the present invention, and these modifications, increase, substitute and therefore be considered in scope of the present invention as defined by the following claims.
Claims (10)
1. system comprises:
Calling terminal in the telecommunication system and terminal called;
In appointment name of before initiating, importing or the preferred ans back at least one in described calling terminal place to the calling of described terminal called;
In operation, described calling terminal is coupled at least one telecommunication system of described terminal called; And
Described appointment name and/or described preferred ans back are the caller information from the calling of calling party at described terminal called place.
2. system according to claim 1, wherein said calling terminal is one of personal computer, subscriber phone, pay phone, VoIP (voice over the Internet protocol) phone or portable terminal, and wherein said terminal called is one of personal computer, subscriber phone, pay phone, voip phone or portable terminal.
3. system according to claim 1, wherein said telecommunication system comprises at least one in PTSN (PSTN) network, internet or the cellular network.
4. system according to claim 1, wherein said system further comprises the call card feature, wherein when call card is activated, provides at least one of specifying in name or the preferred ans back to described telecommunication system.
5. system according to claim 5, wherein before the calling of initiating described terminal called, when described call card is activated to appointment name that described telecommunication system provides or in the preferred ans back at least one can be at described calling terminal place with specify name or preferably new at least one in the ans back cover.
6. system according to claim 1, wherein said calling is voip call.
7. method comprises:
Before the calling of initiating the terminal called in the telecommunication system, in designated call person name or the preferred ans back different with the acquiescence ans back of calling terminal at least one specified; And
Provide specified caller's name and preferred ans back as caller information at described terminal called place.
8. method according to claim 7, wherein:, when described call card is activated, designated call person name and preferred ans back are specified for call card; For the calling of being undertaken by public telephone office, PSTN (PSTN) phone has the specified caller's name of before making a call input and the optional feature of ans back; For the calling of being undertaken by public telephone office, public pstn telephone has the optional feature of specified caller's name of before making a call input and preferred ans back; Call out to PSTN for public telephone office PC, before making a call, PC (personal computer) client is caught specified caller's name and preferred ans back; And for the calling of being undertaken by the residence voip phone, as the part of activation of service, the person's name that provides the designated call and preferred ans back.
9. method according to claim 7, wherein said calling terminal is one of personal computer, subscriber phone, pay phone, VoIP (voice over the Internet protocol) phone or portable terminal, and wherein terminal called is one of personal computer, subscriber phone, pay phone, voip phone or portable terminal; And wherein said communication network comprises in PTSN network, internet or the cellular network at least one.
10. method according to claim 7, wherein said communication network further comprises the call card feature, wherein provides at least one of specifying in name or the preferred ans back to described telecommunication system when call card is activated.
Applications Claiming Priority (2)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| US11/268,921 US20070115928A1 (en) | 2005-11-08 | 2005-11-08 | Network support for enhanced VoIP caller ID |
| US11/268,921 | 2005-11-08 |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN101305588A true CN101305588A (en) | 2008-11-12 |
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Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNA2006800417507A Pending CN101305588A (en) | 2005-11-08 | 2006-11-02 | Network support for enhanced voip caller id |
Country Status (6)
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| US (1) | US20070115928A1 (en) |
| EP (1) | EP1946535A1 (en) |
| JP (1) | JP2009515482A (en) |
| KR (1) | KR20080066945A (en) |
| CN (1) | CN101305588A (en) |
| WO (1) | WO2007056092A1 (en) |
Families Citing this family (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US20080037729A1 (en) | 2006-08-11 | 2008-02-14 | Jaxtr, Inc. | Method And System For Communicating Across Telephone And Data Networks |
| WO2008042990A1 (en) * | 2006-10-03 | 2008-04-10 | Yardarm Technologies, Llc | Methods and systems for providing advanced call services |
| JP5353110B2 (en) * | 2008-08-18 | 2013-11-27 | 沖電気工業株式会社 | Sender information notification system and communication network server |
| US10973059B2 (en) | 2011-07-27 | 2021-04-06 | Vonage America, Llc | Systems and methods of providing communications services |
| US20130121214A1 (en) | 2011-07-27 | 2013-05-16 | Vonage Network, Llc | Systems and methods of providing communications services |
| US8265083B1 (en) | 2011-07-27 | 2012-09-11 | Vonage Network, Llc | Systems and methods of providing communications services |
| CN103209120B (en) * | 2013-03-14 | 2017-04-26 | 中兴通讯股份有限公司 | Gateway device, VoIP (Voice over Internet Portocol) call management and control system and VoIP call management and control method |
| JP5682658B2 (en) * | 2013-06-12 | 2015-03-11 | 沖電気工業株式会社 | Communication device and communication terminal |
| CN104243728A (en) * | 2013-06-24 | 2014-12-24 | 中兴通讯股份有限公司 | Calling and called method of terminal, calling and called terminals and communication method and system |
| KR101685466B1 (en) * | 2014-08-28 | 2016-12-12 | 삼성에스디에스 주식회사 | Method for extending participants of video conference service |
Family Cites Families (11)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US6711402B1 (en) * | 1997-08-06 | 2004-03-23 | Nortel Networks Ltd. | Method and apparatus for displaying calling party information |
| US5903636A (en) * | 1997-10-27 | 1999-05-11 | Bellsouth Corporation | System and method for providing caller identification in conjunction with calling card calls |
| US6498791B2 (en) * | 1998-04-03 | 2002-12-24 | Vertical Networks, Inc. | Systems and methods for multiple mode voice and data communications using intelligently bridged TDM and packet buses and methods for performing telephony and data functions using the same |
| JP2001358847A (en) | 2000-06-12 | 2001-12-26 | Canon Inc | Gateway device |
| US20020080941A1 (en) * | 2000-12-21 | 2002-06-27 | Clapper Edward O. | Message card |
| US20030046575A1 (en) * | 2001-08-30 | 2003-03-06 | International Business Machines Corporation | Digital identity information cards |
| US6757533B2 (en) * | 2002-01-23 | 2004-06-29 | Nokia Corporation | Rich calling line handling in call setup signalling |
| US7533099B2 (en) * | 2004-12-02 | 2009-05-12 | Microsoft Corporation | Componentized remote user interface |
| US20050105705A1 (en) * | 2003-11-14 | 2005-05-19 | General Instrument Corporation | Personalized caller identification |
| US7191919B2 (en) * | 2004-09-01 | 2007-03-20 | Acrison, Inc. | Weight-loss weigh feeder with pressure compensation |
| US7451921B2 (en) * | 2004-09-01 | 2008-11-18 | Eric Morgan Dowling | Methods, smart cards, and systems for providing portable computer, VoIP, and application services |
-
2005
- 2005-11-08 US US11/268,921 patent/US20070115928A1/en not_active Abandoned
-
2006
- 2006-11-02 CN CNA2006800417507A patent/CN101305588A/en active Pending
- 2006-11-02 JP JP2008540081A patent/JP2009515482A/en active Pending
- 2006-11-02 KR KR1020087011053A patent/KR20080066945A/en not_active Application Discontinuation
- 2006-11-02 WO PCT/US2006/042869 patent/WO2007056092A1/en active Application Filing
- 2006-11-02 EP EP06827412A patent/EP1946535A1/en not_active Withdrawn
Also Published As
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| WO2007056092A1 (en) | 2007-05-18 |
| EP1946535A1 (en) | 2008-07-23 |
| JP2009515482A (en) | 2009-04-09 |
| US20070115928A1 (en) | 2007-05-24 |
| KR20080066945A (en) | 2008-07-17 |
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Application publication date: 20081112 |